SIP Profile Configuration

A SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, call pickup URI, and so on. The profiles contain some standard entries that cannot be deleted or changed.

SIP Profile Configuration Settings

A SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, call pickup URI, and so on. The profiles contain some standard entries that cannot be deleted or changed.

Enter a name to identify the SIP profile; for example, SIP_7905. The value can include 1 to 50 characters, including alphanumeric characters, dot, dash, and underscores.

Description

This field identifies the purpose of the SIP profile; for example, SIP for 7970.

Default MTP Telephony Event Payload Type

This field specifies the default payload type for RFC2833 telephony event. See RFC 2833 for more information. In most cases, the default value specifies the appropriate payload type. Be sure that you have a firm understanding of this parameter before changing it, as changes could result in DTMF tones not being received or generated. The default value specifies 101 with range from 96 to 127.

The value of this parameter affects calls with the following conditions:

•For the calling SIP trunk, the Media Termination Point Required check box is checked on the SIP Trunk Configuration window.

Resource Priority Namespace List

Select a configured Resource Priority Namespace Network Domain list from the drop-down menu. Configure the lists in the Resource Priority Namespace Network Domain menu that is accessed from the System menu.

To enable or disable Early Offer for G.Clear Calls, choose one of the following options:

•Disabled

•CLEARMODE

•CCD

•G.nX64

•X-CCD

Redirect by Application

Checking this check box and configuring this SIP Profile on the SIP trunk allows the Cisco Unified Communications Manager administrator to

•Apply a specific calling search space to redirected contacts that are received in the 3xx response.

•Apply digit analysis to the redirected contacts to make sure that the call get routed correctly.

•Prevent DOS attack by limiting the number of redirection (recursive redirection) that a service parameter can set.

•Allow other features to be invoked while the redirection is taking place.

Getting redirected to a restricted phone number (such as an international number) means that handling redirection at the stack level will cause the call to be routed instead of being blocked. This represents the behavior that you will get if the Redirect by Application check box is unchecked.

By default, Cisco Unified Communications Manager will signal the calling phone to play local ringback if SDP is not received in the 180 or 183 response. If SDP is included in the 180 or 183 response, instead of playing ringback locally, Cisco Unified Communications Manager will connect media, and the calling phone will play whatever the called device is sending (such as ringback or busy signal). If you do not receive ringback, the device to which you are connecting may be including SDP in the 180 response, but it is not sending any media before the 200OK response. In this case, check this check box to play local ringback on the calling phone and connect the media upon receipt of the 200OK response

Note Even though the phone that is receiving ringback is the calling phone, you need the configuration on the called device profile because it determines the behavior.

The parameter allows the system to accept a signal from Microsoft Exchange that causes it to switch the call from audio to T.38 fax. To use this feature, you must also configure a SIP trunk with this SIP profile. For more information, see Chapter 92, "Trunk Configuration."

Note The parameter applies to SIP trunks only, not phones that are running SIP or other endpoints.

Enable ANAT

This option allows a dual-stack SIP trunk to offer both IPv4 and IPv6 media.

When you check both the Enable ANAT and the MTP Required check boxes, Cisco Unified Communications Manager inserts a dual-stack MTP and sends out an offer with two m-lines, one for IPv4 and another for IPv6. If a dual- stack MTP cannot be allocated, Cisco Unified Communications Manager sends an INVITE without SDP.

When you check the Enable ANAT check box and the Media Termination Point Required check box is unchecked, Cisco Unified Communications Manager sends an INVITE without SDP.

When the Enable ANAT and Media Termination Point Required check boxes display as unchecked (or when an MTP cannot be allocated), Cisco Unified Communications Manager sends an INVITE without SDP.

When you uncheck the Enable ANAT check box but you check the Media Termination Point Required check box, consider the information, which assumes that an MTP can be allocated:

•Cisco Unified Communications Manager sends an IPv4 address in the SDP for SIP trunks with an IP Addressing Mode of IPv4 Only.

•Cisco Unified Communications Manager sends an IPv6 address in the SDP for SIP trunks with an IP Addressing Mode of IPv6 Only.

•For dual-stack SIP trunks, Cisco Unified Communications Manager determines which IP address type to send in the SDP based on the configuration for the IP Addressing Mode Preference for Media enterprise parameter.

Parameters used in Phone

Timer Invite Expires (seconds)

This field specifies the time, in seconds, after which a SIP INVITE expires. The Expires header uses this value. Valid values include any positive number; 180 specifies the default.

Timer Register Delta (seconds)

Use this parameter in conjunction with the Timer Register Expires setting. The phone will reregister Timer Register Delta seconds before the registration period ends. The registration period gets determined by the value of the SIP Station Keepalive Interval service parameter. Valid values range from 32767 to 0. Default specifies 5.

Timer Register Expires (seconds)

This field specifies the value that the phone that is running SIP will send in the Expires header of the REGISTER message. Valid values include any positive number; however, 3600 (1 hour) specifies the default value. In the 200OK response to REGISTER, Cisco Unified Communications Manager will include an Expires header with the configured value of the SIP Station KeepAlive Interval service parameter. This value in the 200OK determines the time, in seconds, after which the registration expires. The phone will refresh the registration Timer Register Delta seconds before the end of this interval.

Timer T1 (msec)

This field specifies the lowest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive number. Default specifies 500.

Timer T2 (msec)

This field specifies the highest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive number. Default specifies 4000.

Retry INVITE

This field specifies the maximum number of times that an INVITE request will be retransmitted. Valid values include any positive number. Default specifies 6.

Retry Non-INVITE

This field specifies the maximum number of times that a SIP message other than an INVITE request will be retransmitted. Valid values include any positive number. Default specifies 10.

If you have a call on hold and are talking on another call, when you hang up the call, this parameter causes the phone to ring to let you know that you still have another party on hold. Valid values follow:

•Off permanently and cannot be turned on and off locally by using the user interface.

•On permanently and cannot be turned on and off locally by using the user interface.

Anonymous Call Block

This field configures anonymous call block. Valid values follow:

•Off—Disabled permanently and cannot be turned on and off locally by using the user interface.

•On—Enabled permanently and cannot be turned on and off locally by using the user interface.

Caller ID Blocking

This field configures caller ID blocking. When blocking is enabled, the phone blocks its own number or e-mail address from phones that have caller identification enabled. Valid values follow:

•Off—Disabled permanently and cannot be turned on and off locally by using the user interface.

•On—Enabled permanently and cannot be turned on and off locally by using the user interface.

Do Not Disturb Control

This field sets the Do Not Disturb (DND) feature. Valid values follow:

•User—The dndControl parameter for the phone should specify 0.

•Admin—The dndControl parameter for the phone should specify 2.

Telnet Level for 7940 and 7960

Cisco Unified IP Phones 7940 and 7960 do not support ssh for login access or HTTP that is used to collect logs; however, these phones support Telnet, which lets the user control the phone, collect debugs, and look at configuration settings. This field controls the telnet_level configuration parameter with the following possible values:

•Disabled (no access)

•Limited (some access but cannot run privileged commands)

•Enabled (full access)

Timer Keep Alive Expires (seconds)

Cisco Unified Communications Manager requires a keepalive mechanism to support redundancy. This field specifies the interval between keepalive messages that are sent to the backup Cisco Unified Communications Manager to ensure that it is available in the event that a failover is required.

Timer Subscribe Expires (seconds)

This field specifies the time, in seconds, after which a subscription expires. This value gets inserted into the Expires header field. Valid values include any positive number; however, 120 specifies the default value.

Timer Subscribe Delta (seconds)

Use this parameter in conjunction with the Timer Subscribe Expires setting. The phone will resubscribe Timer Subscribe Delta seconds before the subscription period ends, as governed by Timer Subscribe Expires. Valid values range from 3 to 15. Default specifies 5.

Maximum Redirections

Use this configuration variable to determine the maximum number of times that the phone will allow a call to be redirected before dropping the call. Default specifies 70 redirections.

Off Hook to First Digit Timer (microseconds)

This field specifies the time in microseconds that passes when the phone goes off hook and the first digit timer gets set. The value ranges from 0 - 15,000 microseconds. Default specifies 15,000 microseconds.

Call Forward URI

This URI provides a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call forward feature.

Abbreviated Dial URI

This URI provides a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the abbreviated dial feature.

Speed dials that are not associated with a line key (abbreviated dial indices) will not download to the phone. The phone will use the feature indication mechanism (INVITE with Call-Info header) to indicate when an abbreviated dial number has been entered. The request URI will contain the abbreviated dial digits (for example, 14), and the Call-Info header will indicate the abbreviated dial feature. Cisco Unified Communications Manager will translate the abbreviated dial digits into the configured digit string and extend the call with that string. If no digit string has been configured for the abbreviated dial digits, a 404 Not Found response gets returned to the phone.

Conference Join Enabled

This check box determines whether the Cisco Unified IP Phones 7940 or 7960, when the conference initiator that is using that phone hangs up, should attempt to join the remaining conference attendees. Check the check box if you want to join the remaining conference attendees; leave it unchecked if you do not want to join the remaining conference attendees.

Note This check box applies to the Cisco Unified IP Phones 7941/61/70/71/11 when they are in SRST mode only.

RFC 2543 Hold

Check this check box to enable setting connection address to 0.0.0.0 per RFC2543 when call hold is signaled to Cisco Unified Communications Manager. This allows backward compatibility with endpoints that do not support RFC3264.

Semi Attended Transfer

This check box determines whether the Cisco Unified IP Phones 7940 and 7960 caller can transfer the second leg of an attended transfer while the call is ringing. Check the check box if you want semi-attended transfer enabled; leave it unchecked if you want semi-attended transfer disabled.

Note This check box applies to the Cisco Unified IP Phones 7941/61/70/71/11 when they are in SRST mode only.

Enable VAD

Check this check box if you want voice activation detection (VAD) enabled; leave it unchecked if you want VAD disabled. When VAD is enabled, no media gets transmitted when voice is detected.

Stutter Message Waiting

Check this check box if you want stutter dial tone when the phone goes off hook and a message is waiting; leave unchecked if you do not want a stutter dial tone when a message is waiting.

This setting supports Cisco Unified IP Phones 7960 and 7940 that run SIP.

Call Stats

Check this check box if you want RTP statistics in BYE requests and responses enabled; leave unchecked if you want RTP statistics in BYE requests and responses disabled.

If this check box is checked, the phone inserts the headers RTP-RxStat and RTP-TxStat as follows:

•RTP-RxStat:Dur=a,Pkt=b,Oct=c,LatePkt=d,LostPkt=e,AvgJit=f

•RTP-TxStat: Dur=g,Pkt=h,Oct=i

where:

•Dur—Total number of seconds since the beginning of reception or transmission.

•Pkt—Total number of RTP packets that are received or transmitted.

•Oct—Total number of RTP payload octets that are received or transmitted (not including RTP header).

•LatePkt—Total number of late RTP packets that are received.

•LostPkt—Total number of lost RTP packets that are received (not including the late RTP packets).

•AvgJit—Average jitter, which is an estimate of the statistical variance of the RTP packet interarrival time, measured in timestamp unit and calculated according to RFC 1889.

•a, b, c, d, e, f, g, h, and i—Integers

Trunk Specific Configuration

Reroute Incoming Request to new Trunk based on

Cisco Unified Communications Manager only accepts calls from the SIP device whose IP address matches the destination address of the configured SIP trunk. In addition, the port on which the SIP message arrives must match the one that is configured on the SIP trunk. After Cisco Unified Communications Manager accepts the call, Cisco Unified Communications Manager uses the configuration for this setting to determine whether the call should get rerouted to another trunk.

From the drop-down list box, choose the method that Cisco Unified Communications Manager uses to identify the SIP trunk where the call gets rerouted:

•Never—If the SIP trunk matches the IP address of the originating device, choose this option, which equals the default setting. Cisco Unified Communications Manager, which identifies the trunk by using the source IP address of the incoming packet and the signaling port number, does not route the call to a different (new) SIP trunk. The call occurs on the SIP trunk on which the call arrived.

•Contact Info Header—If the SIP trunk uses a SIP proxy, choose this option. Cisco Unified Communications Manager parses the contact header in the incoming request and uses the IP address or domain name and signaling port number that is specified in the header to reroute the call to the SIP trunk that uses the IP address and port. If no SIP trunk is identified, the call occurs on the trunk on which the call arrived.

•Call-Info Header with purpose=x-cisco-origIP—If the SIP trunk uses a Customer Voice Portal (CVP) or a Back-to-Back User Agent (B2BUA), choose this option. When the incoming request is received, Cisco Unified Communications Manager parses the Call-Info header, looks for the parameter, purpose=x-cisco-origIP, and uses the IP address or domain name and the signaling port number that is specified in the header to reroute the call to the SIP trunk that uses the IP address and port. If the parameter does not exist in the header or no SIP trunk is identified, the call occurs on the SIP trunk on which the call arrived.

Tip This setting does not work for SIP trunks that are connected to a Cisco Unified Presence proxy server or SIP trunks that are connected to originating gateways in different Cisco Unified CM groups.

Finding SIP Profiles

This topic describes how to use the Find and List SIP Profile window. The function searches every type of SIP profile against the following categories:

•Profile name

•Description

Procedure

Step 1 Choose Device > Device Settings > SIP Profile.

The Find and List SIP Profiles window displays. Records from an active (prior) query may also display in the window.

Step 2 To find all records in the database, ensure the dialog box is empty; go to Step 3.

To filter or search records

•From the first drop-down list box, select a search parameter.

•From the second drop-down list box, select a search pattern.

•Specify the appropriate search text, if applicable.

Note To add additional search criteria, click the + button. When you add criteria, the system searches for a record that matches all criteria that you specify. To remove criteria, click the - button to remove the last added criterion or click the Clear Filter button to remove all added search criteria.

Step 3 Click Find.

All matching records display. You can change the number of items that display on each page by choosing a different value from the Rows per Page drop-down list box.

Note You can delete multiple records from the database by checking the check boxes next to the appropriate record and clicking Delete Selected. You can delete all configurable records for this selection by clicking Select All and then clicking Delete Selected.

Step 4 From the list of records that display, click the link for the record that you want to view.

Note To reverse the sort order, click the up or down arrow, if available, in the list header.

Configuring SIP Profiles

Perform the following procedure to add, copy, or update a SIP profile.

Procedure

Step 1 Choose Device > Device Settings > SIP Profile.

The Find and List SIP Profile window displays.

Step 2 Perform one of the followings tasks:

•To copy an existing SIP profile, locate the appropriate SIP profile as described in "Finding SIP Profiles" section, click the Copy button next to the SIP profile that you want to copy and continue with Step 3.

•To add a new SIP profile, click the Add New button and continue with Step 3.

Deleting SIP Profiles

This section describes how to delete a SIP profile.

Before You Begin

To find out which devices are using the SIP profile, choose Dependency Records link from the Related Links drop-down list box in the SIP Profile Configuration window. If the dependency records are not enabled for the system, the dependency records summary window displays a message. For more information about dependency records, see the "Accessing Dependency Records" section on page A-2.

Synchronizing a SIP Profile With Affected SIP Devices

To synchronize SIP devices with a SIP profile that has undergone configuration changes, perform the following procedure, which will apply any outstanding configuration settings in the least-intrusive manner possible. (For example, a reset/restart may not be required on some affected devices.)

Procedure

Step 1 Choose Device > Device Settings > SIP Profile.

The Find and List SIP Profiles window displays.

Step 2 Choose the search criteria to use.

Step 3 Click Find.

The window displays a list of SIP Profiles that match the search criteria.